Reland "Apply global audio options at the engine level" This is a reland of commit 1488f778a7b25c67c1c30dd9e11f9588d4b2ff38 Reland changes: Move ownership of global audio options to ConnectionContext instead of WebRtcVoiceEngine. That avoids an issue with a downstream implementation where a premature Init/Terminate cycle occurred during CreateAudioSource Original change's description: > Apply global audio options at the engine level > > Introduce ApplyGlobalOptions to the MediaEngine interface to configure > global audio processing options (such as AEC, AGC, and noise > suppression) directly at the engine level. This replaces the previous > behavior where media channels implicitly re-applied options to the > engine during initialization and parameter updates. > > Applying global settings during CreateAudioSource ensures engine-level > configurations persist reliably across stream lifecycles. This change > also separates global audio processing options from stream-level > parameters like jitter buffer and audio network adaptor settings. > > [Before Decoupling] > +----------------------------------------------------+ > | AudioOptions (AEC/AGC/NS + Jitter/ANA) | > +----------------------------------------------------+ > / \ > / (engine()->ApplyOptions) \ > v v > WebRtcVoiceEngine (Global APM) VoiceSend/RecvChannel > > > [After Decoupling (This CL)] > +-----------------------+ +-----------------------+ > | Global Options (APM) | | Channel Options (ST) | > | - echo_cancellation | | - jitter_buffer | > | - auto_gain_control | | - audio_network_ | > | - noise_suppression | | adaptor | > +-----------------------+ +-----------------------+ > | | > v (ApplyGlobalOptions) v > WebRtcVoiceEngine (APM) VoiceSend/RecvChannel > - Stores in global_options_ - No ApplyOptions calls > - Persisted across cycles - No leakage / racing > > Bug: webrtc:42224170 > Change-Id: I3c23293ae338495ee32330ce5287411947f6d472 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/476340 > Reviewed-by: Per Åhgren <peah@webrtc.org> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#47844} Bug: webrtc:42224170 Change-Id: I2e3b0374885c5f8daf25cc2fd62b56653574c13e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/476782 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#47867}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
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Authoritative list of directories that contain the native API header files.